Open-source alternatives to Fastrtc

Pipecat is a framework and software development kit for building real-time multimodal AI agents and speech-to-speech systems. It utilizes a frame-based data pipeline to route audio, video, and text through a modular sequence of processors, enabling the orchestration of low-latency conversational AI. The project is distinguished by its ability to coordinate complex multimodal services, including speech-to-text, language models, and text-to-speech, within a single pipeline. It features semantic voice activity detection for natural turn-taking, state-machine conversation flows for dialogue manag

aiortc is a Python implementation of the WebRTC protocol, providing an asynchronous stack for real-time audio, video, and data streaming between peers. It functions as a media engine and data channel implementation that manages peer connections through a non-blocking framework. The library enables the establishment of direct peer-to-peer networking by negotiating session descriptions and connectivity paths. It facilitates secure, bidirectional data transmission via data channels and supports low-latency media streaming using standard codecs. The system includes capabilities for network NAT t

LiveTalking is an interactive talking head engine and AI avatar management platform designed to synchronize synthetic speech with facial movements. It functions as a real-time orchestrator that connects large language models and text-to-speech services to neural-rendered digital humans. The project distinguishes itself through low-latency streaming capabilities and the ability to handle real-time conversational interruptions. It supports advanced audio-visual customization, including human voice cloning and the ability to drive avatar expressions using real-time webcam data. The platform cov

Vocode-core is a framework for building real-time conversational AI voice agents. It serves as a conversational orchestrator and pipeline that integrates speech-to-text, large language models, and text-to-speech services to enable low-latency voice interactions. The project features a provider-agnostic interface that allows for swappable speech and language model providers, including support for both cloud APIs and local binaries. It distinguishes itself through a specialized telephony integration layer that enables agents to be deployed across phone lines, WebRTC, and virtual meeting platfor

SimpleWebRTC is a communication framework and real-time media streaming library designed to establish peer-to-peer audio, video, and data streams between web clients. It provides a conference room manager to organize multiple participants into virtual rooms for group interaction. The framework includes a dedicated system for peer-to-peer file transfers and low-latency data messaging. It also features a network traversal configuration tool for managing the servers required to maintain connectivity across firewalls and restrictive network environments. The project covers broad capability areas

This project is a WebRTC plugin for Flutter that provides a real-time communication framework for implementing peer-to-peer voice, video, and data streaming. It functions as a cross-platform media streamer and hardware media controller, allowing applications to manage device cameras and microphones across mobile, desktop, and web platforms. The framework includes a dedicated peer-to-peer data channel interface for exchanging arbitrary data packets with low latency. It ensures privacy and data integrity through end-to-end encryption for all audio, video, and data transmissions. Broad capabili

react-native-webrtc is a mobile framework wrapper and peer-to-peer communication library used to integrate real-time audio, video, and data communication into cross-platform mobile applications. It functions as a tool for capturing device screens and streaming live media content via the WebRTC protocol. The project provides an interface for bidirectional data channels, enabling the exchange of arbitrary data packets between peers without a central relay server. It supports real-time media streaming, including peer-to-peer audio and video transmission and screen capture capabilities. The libr

This project is a scalable real-time communication platform designed for multi-party video and audio conferencing. It functions as a modular media server that orchestrates live sessions, manages virtual meeting rooms, and handles the distribution of media streams across distributed computing environments. The platform distinguishes itself through its support for protocol bridging, allowing for the translation between WebRTC, RTSP, SIP, and HLS. It incorporates hardware-accelerated transcoding to maintain performance across diverse devices and utilizes a plugin-based architecture to enable cus

Ant-Media-Server is a server-side streaming engine designed for real-time audio and video broadcasting. It functions as a live media transcoder and WebRTC streaming server, facilitating the ingestion, processing, and delivery of media streams across multiple protocols. The platform is distinguished by its cloud-scale media infrastructure, which automates server capacity expansion to handle fluctuating volumes of concurrent viewers. It utilizes adaptive bitrate streaming to adjust video quality in real time based on network conditions and employs WebRTC and SRT to achieve ultra-low latency del

This is a WebRTC video calling platform and browser-based video conferencing tool. It provides an instant communication room generator and an embeddable video call widget for hosting group audio and video calls directly in the browser. The platform enables a no-authentication communication workflow, allowing for the creation of unique, shareable links to establish private video sessions without requiring user account registration or software installation. The system supports real-time group video calling and provides capabilities for integrating the communication interface into external web

This project is a real-time communication and media streaming server designed for broadcasting, recording, and distributing audio and video content. It functions as a live streaming server and an RTMP media server, providing the infrastructure necessary to deliver real-time media to multiple concurrent viewers over a network. The system distinguishes itself through a multi-protocol media gateway that supports RTMP, E-RTMP v2 for modern codecs like HEVC and AV1, and HTTP media tunneling to bypass restrictive firewalls. It further acts as a real-time communication platform by synchronizing shar

This library provides a complete implementation of the WebRTC protocol suite in Rust, enabling peer-to-peer audio, video, and data channels with an asynchronous, Rust-native stack. Its architecture separates protocol logic from I/O, giving developers control over threading and scheduling while the library manages the protocol state. The stack is built to be modular, supporting data channel multiplexing over a single SCTP association, an event-driven callback interface for responding to connection changes, and an interceptor pipeline for processing media packets without altering core protocol c

This repository provides a collection of reference implementations and practical demonstrations for using WebRTC to establish real-time audio, video, and data communication. It contains code samples for negotiating peer-to-peer connections, managing media streams, and utilizing low-latency data channels. The project demonstrates the capture of audio and video from hardware devices, as well as the redirection of canvas element content into media streams. It includes examples of transferring arbitrary text and binary data between peers and managing the negotiation of direct connections. The sa

Duix-Mobile is a software development kit for deploying real-time conversational AI characters on mobile devices. It enables the creation of interactive digital humans capable of fluid voice-to-voice interactions, featuring low-latency speech recognition and synchronized lip movements. The project distinguishes itself through the ability to integrate custom external language models and speech providers to define an avatar's intelligence and voice. It supports the generation of real-time multilingual subtitles and provides mechanisms to track the training status of newly created digital charac

This project is a collection of technical demonstrations and prototypes that showcase various HTML5 APIs, multimedia capabilities, and web standard implementations. It serves as a frontend technology sandbox and a gallery of implementation proofs-of-concept for modern browser features. The collection includes specialized showcases for HTML5 multimedia, featuring the manipulation of audio, video, and real-time media streams. It also provides a series of experiments focused on prototyping web capabilities to verify technical feasibility and browser support in live environments. The project cov

This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing, specifically engineered for Apple Silicon. It provides a framework for building real-time, full-duplex voice agents that operate entirely offline, leveraging native hardware acceleration to maintain performance and privacy. By utilizing optimized machine learning models, the library enables local execution of complex audio tasks without reliance on external cloud services. The library distinguishes itself through its specialized focus on local, high-performance voice interaction. It incl

Ten Framework is a multimodal large language model agent framework designed for building low-latency conversational agents. It integrates voice, text, and visual inputs in real time to facilitate human interaction. The project includes a real-time speech processing pipeline for streaming transcription, voice activity detection, and speaker diarization. It also features an avatar synchronization engine that coordinates character lip animations and visual outputs with synthesized speech. The framework covers edge AI deployment through containerized packaging and direct integration with embedde

simple-peer is a JavaScript library that provides a wrapper for WebRTC to simplify the establishment of peer-to-peer networking in the browser. It serves as a tool for creating direct device-to-device connections for the transmission of binary data and real-time media streaming. The library manages the exchange of strings and binary buffers through a data channel implementation and provides tools for sharing real-time audio and video tracks between peers. It covers the full lifecycle of peer connectivity, including signaling coordination, session description negotiation, and the gathering of

Project-Lightspeed is a low-latency video relay and WebRTC live streaming server. It functions as an OBS streaming backend that ingests media streams and converts them for real-time playback in web browsers. The system operates as a WebSocket stream orchestrator, managing connection handshakes and stream authentication to coordinate signaling between the server and the client browser. It utilizes a high-speed protocol to achieve sub-second delay between the broadcast source and the viewer. The server covers media relay capabilities, including network packet relaying and the ingestion of vide

This project is a framework for building local voice assistants and a real-time audio streaming server. It functions as a containerized inference engine and a multilingual speech pipeline that orchestrates speech-to-text, language models, and text-to-speech components to convert spoken input into spoken output. The system is distinguished by its use of WebSocket-based bidirectional streaming for low-latency interactions. It features a voice activity detection system that manages speech boundaries and handles user barge-in interruptions during assistant playback. It also supports custom voice

Jitsi is an open-source communication suite providing a self-hosted platform for real-time video conferencing, audio calls, and instant messaging. It functions as a comprehensive toolkit for streaming voice and video across different devices and platforms, allowing for the deployment of private team collaboration environments. The project is distinguished by its multi-protocol communication bridging, which translates and exchanges data between diverse network standards including SIP, XMPP, and IRC. This allows users on different protocols to interact within a unified communication interface.

flv.js is a JavaScript library and HTML5 Media Source Extension wrapper designed to play FLV video and live streams in web browsers. It enables the rendering of FLV content within a standard web video element without the need for external plugins. The project focuses on real-time transmuxing, converting FLV container data into fragmented MP4 segments to ensure browser compatibility. It includes specific implementations for low-latency live streaming and cross-origin video loading via CORS-compliant server headers. The library provides capabilities for segmented media playback, media position

Sherpa-ONNX is an ONNX-based speech processing toolkit that provides a local speech recognition engine, an on-device voice synthesis tool, and a speaker identification framework. It is designed as a cross-platform speech API that enables speech-to-text, text-to-speech, and speaker verification tasks to be executed locally on a device without requiring network access. The project is distinguished by its ability to perform zero-shot voice cloning and speaker diarization on-device. It supports a wide range of hardware accelerations, including GPU and various NPU architectures, and provides a Web

This project is a comprehensive reference guide and directory of web browser capabilities. It serves as a technical map for accessing native operating system functions, hardware interfaces, and standard web APIs to bridge the gap between web applications and desktop or mobile environments. The resource provides detailed guidance on implementing Progressive Web App features, including offline caching, push notifications, and native installation prompts. It also catalogs methods for interacting with hardware peripherals via USB, Bluetooth, and NFC, as well as reading raw data from device sensor

SRS is a real-time media server designed to ingest, route, and distribute live audio and video streams across various transport protocols. It functions as a multi-protocol stream relay, including a dedicated RTMP media gateway and a WebRTC signaling server to coordinate peer-to-peer media exchanges. The system features a multi-protocol relay engine that transforms incoming media packets between different transport formats without re-encoding. This allows it to serve as a video delivery proxy that routes live media from a single source to multiple concurrent viewers using diverse delivery prot

VidGear is a high-performance Python video processing framework designed for capturing, transcoding, and manipulating video streams. It functions as a multi-protocol video streamer and a WebRTC streaming server, enabling the transfer of video frames over networks using RTSP, RTMP, RTP, and MJPEG protocols. The project distinguishes itself through hardware-accelerated video transcoding and decoding using GPU backends like CUDA to reduce CPU load. It includes a cross-platform screen capture tool and a specialized system for establishing direct peer-to-peer media connections using WebRTC signali